* suggested evolution (for the dumbs possibly )
- at boot, error message if ethernet unavailable
- at boot, error message if remote file server unavailable
- option to display the cover art on the device (eg I use a NetBoot as my server with Mpod remote control but would find it great to have a permanent display with a large size)
- use a rating for titles/album in the mpd database

* more general question : i use Mpod and Mpad on Apple devices but find these clients very limited, specifically the display of long titles or album names. What are good alternatives ?

Hi!
I have been using MPD puppy for some time, and I am very happy for the labor that has been put into this software. Thank you.

I have a question. I don´t know but maybe the information about this are to be find in the forum somewhere, but there is a lot of posts to read through.
Anyway. My question is simple. If I play a - say 24 bit 96 kHz file. Do MPD pup any resampling to 16 bit 44,1? Or is the file sent to the DAC in it´s original bit and samplerate? I am asking, because when running my DAC under Windows it only accepts maximum 16 bits and 48 kHz. Running under MPD puppy there is no problem playing 24 bits 96 kHz, but is this due to resampling taking place?

@ Jean-louis - I'll try to implement some of those suggestions - some straightforward, others are easier said than done.

Regarding MPoD/MPaD, I'm not aware of any good alternatives for iOS. Best option might be to try some of the PHP clients in the iOS Safari browser.

Magellan wrote:

I have a question. I don´t know but maybe the information about this are to be find in the forum somewhere, but there is a lot of posts to read through.
Anyway. My question is simple. If I play a - say 24 bit 96 kHz file. Do MPD pup any resampling to 16 bit 44,1? Or is the file sent to the DAC in it´s original bit and samplerate? I am asking, because when running my DAC under Windows it only accepts maximum 16 bits and 48 kHz. Running under MPD puppy there is no problem playing 24 bits 96 kHz, but is this due to resampling taking place?

Hi, glad to hear it's been working for you. The behavior depends on your Sound device/DAC. When it's initialized it tells ALSA what bitrates it supports. If you play a bitrate that the DAC doesn't support then it will be resampled to a bitrate that your DAC does support. This behavior can be disabled, but if you do that it means you won't be able to play back content at bitrates your DAC doesn't support.Last edited by ldolse on Thu 31 Jan 2013, 07:56; edited 1 time in total

I have a question. I don´t know but maybe the information about this are to be find in the forum somewhere, but there is a lot of posts to read through.
Anyway. My question is simple. If I play a - say 24 bit 96 kHz file. Do MPD pup any resampling to 16 bit 44,1? Or is the file sent to the DAC in it´s original bit and samplerate? I am asking, because when running my DAC under Windows it only accepts maximum 16 bits and 48 kHz. Running under MPD puppy there is no problem playing 24 bits 96 kHz, but is this due to resampling taking place?

Best Regards

Hi, glad to hear it's been working for you. The behavior depends on your Sound device/DAC. When it's initialized it tells ALSA what bitrates it supports. If you play a bitrate that the DAC doesn't support then it will be resampled to a bitrate that your DAC does support. This behavior can be disabled, but if you do that it means you won't be able to play back content at bitrates your DAC doesn't support.

My USB DAC also accepts SPDIF. I know it accepts 24 bit 192 kHz over SPDIF. I am not sure however, if it also accepts the same bit- and samplerate over USB. I guess the most simple way to test this is to disable the resampling function in ALSA. Is this covered in the graphic configuration tool, or do I have to perform some command line exercise?

My USB DAC also accepts SPDIF. I know it accepts 24 bit 192 kHz over SPDIF. I am not sure however, if it also accepts the same bit- and samplerate over USB. I guess the most simple way to test this is to disable the resampling function in ALSA. Is this covered in the graphic configuration tool, or do I have to perform some command line exercise?

16/48 over USB is a strange limitation if the SPDIF supports up to 24/192. Usually the limitations are based on what class of USB audio the DAC supports - Class 1 is limited to 24/96, Class 2 can go up to 24/192. 16/48 is generally only seen for really inexpensive/old sound devices which were designed for computer only usage (as 16/48 has historically been the default computer bitrate)

Disabling auto-resampling isn't covered in the Wizard, as it's not something a user would typically do. It would probably require editing some ALSA files as well as mpd.conf, I'm not 100% sure on what the steps would be. It would probably be easier to just Google the model of your DAC and see what the USB interface supports.

My USB DAC also accepts SPDIF. I know it accepts 24 bit 192 kHz over SPDIF. I am not sure however, if it also accepts the same bit- and samplerate over USB. I guess the most simple way to test this is to disable the resampling function in ALSA. Is this covered in the graphic configuration tool, or do I have to perform some command line exercise?

16/48 over USB is a strange limitation if the SPDIF supports up to 24/192. Usually the limitations are based on what class of USB audio the DAC supports - Class 1 is limited to 24/96, Class 2 can go up to 24/192. 16/48 is generally only seen for really inexpensive/old sound devices which were designed for computer only usage (as 16/48 has historically been the default computer bitrate)

Disabling auto-resampling isn't covered in the Wizard, as it's not something a user would typically do. It would probably require editing some ALSA files as well as mpd.conf, I'm not 100% sure on what the steps would be. It would probably be easier to just Google the model of your DAC and see what the USB interface supports.

Info on mpd.conf edits here: http://mpd.wikia.com/wiki/Tuning

The problem is my DAC is built by a one man company, a very small company. It is built into a preamp, and it sounds wonderful, but it is not possible to get any relevant info about it over the internet. I know however that it´s DAC chip is a Cirrus Logic CS4398 stereo audio 24-bit/192 kHz digital/analog converter system.
I use MPod as a client, and it says the FLAC file is 96 kHz during playback, but I don´t know if it is reading the input or output.
Is there a terminal command which can display what samplerate and bitrate are put into the dac?

Is there a terminal command which can display what samplerate and bitrate are put into the dac?

I'm not sure if you can get it with your card. You can try a command like this:

Code:

cat /proc/asound/card1/pcm0p/sub0/hw_params

But you'll need to change card1/pcm0p/sub0 to match the hardware of your actual card. Double-tap the tab key after each slash to see what options are available to you - running cat on the different items will provide you with a variety of debug information about your card. Some of this level of debugging is disabled in mpdPup's kernel, but a lot of detail is still there.

I know we have all been shooting tweaks etc at you...here is an easy way to pretty much get every Linux audio player tweak.

Download Soundchcks squeezebox touch tool box on a linux machine and open it up with a text editor.

You will find not only the prioritization tweaks I use but network tweaks and kernel tweaks etc. No need to try to re-invent the wheel.

Personally I don't like kernel tweaks and network tweaks seem to have a very minimal benefit. But either way pretty much everything is there.

That way you can incorporate all or some into Mpdpup and let a small group of testers, test out the sound and benefit. If your ears and your testers ears give a certain tweak the thumbs up then its a go. Better yet if you have time or the inclination you can allow users to toggle them on/off or choose values during set-up.

I would welcome the inclusion of some of the Linux utils that are required for Soundchck's Toolbox tweaks in the mpdPup build, even if ldolse has not got the time to include the actual tweaks himself in the set-up wizard.

Go ahead and download the file, then open it up with a text editor. You can actually cut and paste some of it in the script you put in the Mpdpup.

However before we go too far with this I would like to Publicly thank Soundcheck for his time and efforts with compiling all these tweaks. Soundcheck has spent a long time working with Linux and deserves all the credit.

Have to say that I found many or most of Soundcheck's code to be very Touch specific. Hardly surprising there. But so much so, that I was finding difficulty in finding any code that would transfer safely or effectively for mpdPup, but have to admit that I am no Linux expert. No doubt ldolse can comment.

ldolse has produced an excellent slimmed down targeted Linux build; it is inevitably not going to have the same kernel or approach as Touch code and therefore will not transfer. Please correct me if I am wrong. Even better point all of us to portions of Touch Toolbox that would just copy & paste across and work 'out of the (tool)box'. I assume that you have done so yourself.

Thanks as always for your suggestions and I am also a fan of Soundcheck in many of his excellent postings on all sorts of hi-fi adventures.

Hi all, i've got a question for Idolse: did you think that is possible to feed some basic commands (such track skip, stop, play and eventually read some informations) to mpd via a rs-232 serial port? For my dac i'm building an mcu to control some function and to have ir remote control and since the alix board has a serial port this idea hitted me, the mcu i'm using can output to rs232, even a web site with some infos can be useful.

Another question, since i don't use it ATM, can i disable albumbler or the database updating it is doing at startup?

Hi all
What would be the best setup for MPDPUP?
I have tryed:
1-Local storage with SSD, just usb DAC connected
2-Network Wifi, server-player, storage on SSD server
3-Network Ethernet server-player, storage on SSD server
4-Network Ethernet server-player, storage on server external USB drive
By far, option 1 gave the best results, bigger soundstage, dynamics and musician identification. The worst was with wifi, noisy and stops on HD files, almost unlistenable.
Have not tryed other minimalistic options for player hardware. They make sense and would like to know your opinion/test results. How does cpu and memory performance affects sound? high/low cost gear?
Best regards

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